/* GStreamer * Copyright (C) 2021 Seungha Yang * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H #include #endif #include "gstasiosrc.h" #include "gstasioobject.h" #include "gstasioringbuffer.h" #include #include #include #include GST_DEBUG_CATEGORY_STATIC (gst_asio_src_debug); #define GST_CAT_DEFAULT gst_asio_src_debug static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_ASIO_STATIC_CAPS)); enum { PROP_0, PROP_DEVICE_CLSID, PROP_CAPTURE_CHANNELS, PROP_BUFFER_SIZE, PROP_OCCUPY_ALL_CHANNELS, PROP_LOOPBACK, }; #define DEFAULT_BUFFER_SIZE 0 #define DEFAULT_OCCUPY_ALL_CHANNELS TRUE #define DEFAULT_LOOPBACK FALSE struct _GstAsioSrc { GstAudioSrc parent; /* properties */ gchar *device_clsid; gchar *capture_channles; guint buffer_size; gboolean occupy_all_channels; gboolean loopback; }; static void gst_asio_src_finalize (GObject * object); static void gst_asio_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_asio_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstCaps *gst_asio_src_get_caps (GstBaseSrc * src, GstCaps * filter); static GstAudioRingBuffer *gst_asio_src_create_ringbuffer (GstAudioBaseSrc * src); #define gst_asio_src_parent_class parent_class G_DEFINE_TYPE (GstAsioSrc, gst_asio_src, GST_TYPE_AUDIO_BASE_SRC); static void gst_asio_src_class_init (GstAsioSrcClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstBaseSrcClass *basesrc_class = GST_BASE_SRC_CLASS (klass); GstAudioBaseSrcClass *audiobasesrc_class = GST_AUDIO_BASE_SRC_CLASS (klass); gobject_class->finalize = gst_asio_src_finalize; gobject_class->set_property = gst_asio_src_set_property; gobject_class->get_property = gst_asio_src_get_property; g_object_class_install_property (gobject_class, PROP_DEVICE_CLSID, g_param_spec_string ("device-clsid", "Device CLSID", "ASIO device CLSID as a string", NULL, (GParamFlags) (GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS))); g_object_class_install_property (gobject_class, PROP_CAPTURE_CHANNELS, g_param_spec_string ("input-channels", "Input Channels", "Comma-separated list of ASIO channels to capture", NULL, (GParamFlags) (GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS))); g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE, g_param_spec_uint ("buffer-size", "Buffer Size", "Preferred buffer size (0 for default)", 0, G_MAXINT32, DEFAULT_BUFFER_SIZE, (GParamFlags) (GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS))); g_object_class_install_property (gobject_class, PROP_OCCUPY_ALL_CHANNELS, g_param_spec_boolean ("occupy-all-channels", "Occupy All Channles", "When enabled, ASIO device will allocate resources for all in/output " "channles", DEFAULT_OCCUPY_ALL_CHANNELS, (GParamFlags) (GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS))); g_object_class_install_property (gobject_class, PROP_LOOPBACK, g_param_spec_boolean ("loopback", "Loopback recording", "Open the sink device for loopback recording", DEFAULT_LOOPBACK, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS))); gst_element_class_add_static_pad_template (element_class, &src_template); gst_element_class_set_static_metadata (element_class, "AsioSrc", "Source/Audio/Hardware", "Stream audio from an audio capture device through ASIO", "Seungha Yang "); basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_asio_src_get_caps); audiobasesrc_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_asio_src_create_ringbuffer); GST_DEBUG_CATEGORY_INIT (gst_asio_src_debug, "asiosrc", 0, "asiosrc"); } static void gst_asio_src_init (GstAsioSrc * self) { self->buffer_size = DEFAULT_BUFFER_SIZE; self->occupy_all_channels = DEFAULT_OCCUPY_ALL_CHANNELS; self->loopback = DEFAULT_LOOPBACK; } static void gst_asio_src_finalize (GObject * object) { GstAsioSrc *self = GST_ASIO_SRC (object); g_free (self->device_clsid); g_free (self->capture_channles); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_asio_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAsioSrc *self = GST_ASIO_SRC (object); switch (prop_id) { case PROP_DEVICE_CLSID: g_free (self->device_clsid); self->device_clsid = g_value_dup_string (value); break; case PROP_CAPTURE_CHANNELS: g_free (self->capture_channles); self->capture_channles = g_value_dup_string (value); break; case PROP_BUFFER_SIZE: self->buffer_size = g_value_get_uint (value); break; case PROP_OCCUPY_ALL_CHANNELS: self->occupy_all_channels = g_value_get_boolean (value); break; case PROP_LOOPBACK: self->loopback = g_value_get_boolean (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_asio_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAsioSrc *self = GST_ASIO_SRC (object); switch (prop_id) { case PROP_DEVICE_CLSID: g_value_set_string (value, self->device_clsid); break; case PROP_CAPTURE_CHANNELS: g_value_set_string (value, self->capture_channles); break; case PROP_BUFFER_SIZE: g_value_set_uint (value, self->buffer_size); break; case PROP_OCCUPY_ALL_CHANNELS: g_value_set_boolean (value, self->occupy_all_channels); break; case PROP_LOOPBACK: g_value_set_boolean (value, self->loopback); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstCaps * gst_asio_src_get_caps (GstBaseSrc * src, GstCaps * filter) { GstAudioBaseSrc *asrc = GST_AUDIO_BASE_SRC (src); GstAsioSrc *self = GST_ASIO_SRC (src); GstCaps *caps = nullptr; if (asrc->ringbuffer) caps = gst_asio_ring_buffer_get_caps (GST_ASIO_RING_BUFFER (asrc->ringbuffer)); if (!caps) caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (src)); if (filter) { GstCaps *filtered = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); caps = filtered; } GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps); return caps; } static GstAudioRingBuffer * gst_asio_src_create_ringbuffer (GstAudioBaseSrc * src) { GstAsioSrc *self = GST_ASIO_SRC (src); GstAsioRingBuffer *ringbuffer = nullptr; HRESULT hr; CLSID clsid = GUID_NULL; GList *device_infos = nullptr; GstAsioDeviceInfo *info = nullptr; GstAsioObject *asio_object = nullptr; glong max_input_ch = 0; glong max_output_ch = 0; std::set < guint > channel_list; std::vector < guint > channel_indices; guint i; gchar *ringbuffer_name; USES_CONVERSION; GST_DEBUG_OBJECT (self, "Create ringbuffer"); if (gst_asio_enum (&device_infos) == 0) { GST_WARNING_OBJECT (self, "No available ASIO devices"); return nullptr; } if (self->device_clsid) { hr = CLSIDFromString (A2COLE (self->device_clsid), &clsid); if (FAILED (hr)) { GST_WARNING_OBJECT (self, "Failed to convert %s to CLSID", self->device_clsid); clsid = GUID_NULL; } } /* Pick the first device */ if (clsid == GUID_NULL) { info = (GstAsioDeviceInfo *) device_infos->data; } else { /* Find matching device */ GList *iter; for (iter = device_infos; iter; iter = g_list_next (iter)) { GstAsioDeviceInfo *tmp = (GstAsioDeviceInfo *) iter->data; if (tmp->clsid == clsid) { info = tmp; break; } } } if (!info) { GST_WARNING_OBJECT (self, "Failed to find matching device"); goto out; } asio_object = gst_asio_object_new (info, self->occupy_all_channels); if (!asio_object) { GST_WARNING_OBJECT (self, "Failed to create ASIO object"); goto out; } /* Configure channels to use */ if (!gst_asio_object_get_max_num_channels (asio_object, &max_input_ch, &max_output_ch) || max_input_ch <= 0) { GST_WARNING_OBJECT (self, "No available input channels"); goto out; } /* Check if user requested specific channel(s) */ if (self->capture_channles) { gchar **ch; ch = g_strsplit (self->capture_channles, ",", 0); auto num_channels = g_strv_length (ch); if (num_channels > max_input_ch) { GST_WARNING_OBJECT (self, "To many channels %d were requested", num_channels); } else { for (i = 0; i < num_channels; i++) { guint64 c = g_ascii_strtoull (ch[i], nullptr, 0); if (c >= (guint64) max_input_ch) { GST_WARNING_OBJECT (self, "Invalid channel index"); channel_list.clear (); break; } channel_list.insert ((guint) c); } } g_strfreev (ch); } if (channel_list.size () == 0) { for (i = 0; i < max_input_ch; i++) channel_indices.push_back (i); } else { for (auto iter : channel_list) channel_indices.push_back (iter); } ringbuffer_name = g_strdup_printf ("%s-asioringbuffer", GST_OBJECT_NAME (src)); ringbuffer = (GstAsioRingBuffer *) gst_asio_ring_buffer_new (asio_object, self->loopback ? GST_ASIO_DEVICE_CLASS_LOOPBACK_CAPTURE : GST_ASIO_DEVICE_CLASS_CAPTURE, ringbuffer_name); g_free (ringbuffer_name); if (!ringbuffer) { GST_WARNING_OBJECT (self, "Couldn't create ringbuffer object"); goto out; } if (!gst_asio_ring_buffer_configure (ringbuffer, channel_indices.data (), channel_indices.size (), self->buffer_size)) { GST_WARNING_OBJECT (self, "Failed to configure ringbuffer"); gst_clear_object (&ringbuffer); goto out; } out: if (device_infos) g_list_free_full (device_infos, (GDestroyNotify) gst_asio_device_info_free); gst_clear_object (&asio_object); return GST_AUDIO_RING_BUFFER_CAST (ringbuffer); }